asterisk users - May 2004

Monday May 31 2004
11:59PM 3 G.729 fallback
10:34PM 1 Failover: iconnecthere to voicepulse
8:47PM 1 zapras how to
7:29PM 2 Billing and CDR's
7:07PM 1 Asterisk and SER Setup Questions.
6:11PM 2 scandsp, voicetronix and rxfx
5:22PM 0 Fedora Core Soft Phone that works?
5:03PM 1 audio problems between asterisk and Cisco 7910 using SCCP
4:18PM 4 wake-up call
3:13PM 0 Fwd: [Serusers] CDR mediation for VoIP
3:04PM 1 Firefly / LibIAX2
2:56PM 0 digium card fax detect AND spandsp
2:51PM 0 MGCP RFC 3015
1:39PM 0 SIP auth in/outside of nat?
12:53PM 0 Import Master.csv in the cdr_mysql database
10:02AM 1 line config
9:38AM 1 I want to purchase atleast one used quicknet card
9:07AM 2 Crc4 issues
8:23AM 0 Disclaimer fax number?
8:11AM 1 Chan Capi Audio Quality Issue...
7:44AM 5 D-Channel Problems
7:08AM 1 Where is my normal dialtone? With DLINK DG-104S (MGCP)
6:54AM 1 * on Opteron
6:21AM 2 Meetme + Billing
6:15AM 6 Asterisk and Zaptel for 2.6 kernel
6:12AM 2 Users in MySQL
5:32AM 1 (no subject)
3:48AM 1 Updated Zaptel this morning and *BOOM* *CRASH*
2:51AM 3 Quicknet PhoneJack Configuration
1:49AM 4 Need guides on setting up PDA on asterisk server
12:59AM 1 Quicknet PhoneJack Configuration files
Sunday May 30 2004
8:08PM 3 spandsp w/libtiff-3.6.1?
6:05PM 1 Advanced access control and cause handling
6:01PM 11 New Firefly version
5:48PM 1 Fwd: hdlc and kernel 2.4.26
3:41PM 0 Codec details in CDR?
12:35PM 5 Unblocking incoming SIP
12:01PM 4 Sipura-spa2000
9:47AM 2 Distribution of Linux
8:00AM 1 G729 Beta Codec
5:48AM 0 bri-stuff.0.0.2
5:41AM 6 Compiling Asterisk with gcc 3.4
2:32AM 6 *** Asterisk Sunday News: Gone Fishing...
1:47AM 1 MacOS X softphone IAX clients?
Saturday May 29 2004
9:51PM 2 Snom and multiple lines
8:32PM 0 Updates
6:49PM 1 Webmin Module in download directory
6:42PM 0 Asterisk - Zaptel - DIGIUM x 4 T1
6:00PM 1 iConnectHere broken?
4:01PM 5 extracting country code from a number
12:25PM 1 SIP extension
11:12AM 3 Odd behaviour with "asterisk -rx"
11:04AM 4 PlayTones problem
9:01AM 1 Delay when routing PSTN -> IAXy dect phone
7:45AM 2 Caller ID with DID
6:16AM 1 transfer bug (#701 -> remote party hears alison, not me)
12:49AM 1 Example: Caller*ID Fixup Macro for use with DIDs
Friday May 28 2004
10:55PM 0 PacketCable with asterisk?
9:20PM 0 HDLC kernel 2.4.26
6:20PM 0 Problem with digits blending on inbound puls ed digits?
4:44PM 2 Sipura SPA-3000 reviews?
2:50PM 0 Problem with digits blending on inbound pulsed digits?
2:16PM 1 TDM31B and Zaptel: FXO port not recognized?
2:11PM 0 Help ! Echo on sip call.
2:11PM 0 Not call pickup for call to sip from mgcp phone
1:48PM 1 [Fwd: Re: call pickup fails.]
1:08PM 6 Beep Sound
12:04PM 0 SIP 404 error....
11:57AM 3 Disable blind xfer
11:51AM 4 PCI 2.2 ??
11:47AM 1 * will not load, after latest CVS install
11:13AM 1 [is this appropriate?] Anyone work for voicepulse or have experience with them
9:43AM 0 seeking an example for Message Waiting Indicator stutter dialtone
9:32AM 16 Asterisk Receptionist manager program.
9:29AM 0 sip client can dial with no registration
9:24AM 0 memory error? TE405P problem?
9:17AM 1 Zap callgroup/pickupgroup question
8:52AM 1 Fw: Asterisk and MySQL
8:31AM 1 Problems with PPP internet T1
8:10AM 11 Asterisk Database
7:43AM 0 E1 channel bank problem
7:17AM 5 Time to lock down v1.1?
6:34AM 9 * as pri_net?
6:27AM 2 spandsp wont compile.
6:13AM 5 Asterisk and MySQL
6:11AM 0 FreeBSD admins * PLEASE HELP!
5:52AM 0 No Sound Card and No Sound from Phone
5:18AM 1 Immortal SIP & NAT problem
5:08AM 1 JTAPI Interface in Asterisk
4:44AM 0 Call transfering
3:52AM 4 Wiki TOS - worrying for an open source project?
3:19AM 5 SIP Changes???
2:49AM 1 asterisk console messages
2:37AM 2 Asterisk with Draytek 2600V
2:16AM 3 Asterisk addons
1:48AM 3 2 Avm fritz passive card in the same box
1:19AM 0 dialogic was RE: "Glare" condition - How well does asteriskhandle?
Thursday May 27 2004
9:41PM 0 mysql-vm-routines does not use the context properly
8:57PM 0 Hangup problem during intergration with 3rd party pbx
8:28PM 1 New to Asterisk - 2 question
8:22PM 0 No stutter MWI on zaptel channel with message waiting
8:11PM 0 Asterisk RPMS Updated (0.9.0 for RH73,8,9 and FC1)
7:36PM 0 HFC-S BRI Slack9.1 kernel 2.6.6 "Guide" bri-stuff.0.0.2
4:50PM 0 [OT] spandsp hylafax asterisk and confusion
2:53PM 0 seeking H.323 <-> MGCP (User Agent) gateway
2:32PM 4 AGI Pascal
2:25PM 0 Re: Asterisk-Users digest, Vol 1 #3921 - 15 msgs
2:21PM 0 Billing, Radius, anyone?
1:26PM 2 Scroll mode in cli
1:01PM 1 Holding and call parking idiosyncrasies...
12:45PM 0 Cisco 7940/60 sip downloads
12:26PM 5 Silly incoming SIP failure
11:20AM 0 Zaptel, analog phone, and call waiting
11:12AM 0 400 Bad request??
10:59AM 1 cvs problem with TDM04B ?
10:11AM 1 Dlink DG-104s telnet reboot
10:07AM 0 WIKI up again
8:55AM 1 Queue Hold Time
8:53AM 3 generate dial tone
8:21AM 2 Asterisk and PostgreSQL
7:44AM 0 Save voice data from IAX. Is that posible?
7:32AM 5 FireFly doesn't work with 3rd party anymore
7:04AM 0 zaphfc: All DTMF tones are doubled
6:32AM 3 dialogic was RE: "Glare" condition - How well does asteriskhandle?
6:30AM 0 threewaycalling
6:04AM 0 return from call parking...
6:02AM 1 What a Difference a NIC makes
5:58AM 1 opinions on as asterisk provider
5:49AM 6 CAPI / Channels
5:44AM 4 Wiki down
5:43AM 0 Simple call generator (shell script)
4:54AM 0 asterisk-oh323, new version 0.6.2
4:46AM 0 bri-stuff-0.0.2 ported to actual development CVS (head)
3:20AM 0 Does anyone know a manager-phone working with asterisk
2:49AM 0 Changing IAX2 refresh times
1:27AM 2 Freenet iPhone w/Asterisk
1:01AM 1 Astersik and PostgreSQL
12:00AM 1 call pickup fails.
Wednesday May 26 2004
11:41PM 1 dialplan AGI DTMF
11:14PM 2 Voicetronix OpenLine4 -- Help Needed
10:57PM 0 quicknet jack card
9:18PM 1 BudgeTone IP phone in Hong Kong
8:13PM 2 Help! No stutter dialtone on message waiting - zaptel phones
5:55PM 0 modprobe wcfxo hangs server
4:39PM 0 Echo on sip to sip call
4:38PM 2 VOIP Service Providers
4:26PM 1 ztdummy with kernel 2.6
2:18PM 1 sip_reg_timeout problem
2:17PM 0 Voicemail Recordings
1:33PM 1 intercept ringing phone
12:36PM 0 Asterisk / SER or both
12:11PM 4 Sipura stun settings
12:06PM 2 SPAM MESSAGE - [Asterisk-Dev] warning message (sound card) - when I run asterisk!!!
11:11AM 2 tieline digit timeout
11:02AM 2 Monitoring Calls
9:30AM 2 tdm04b stopped taking inbound calls - todays cvs
9:26AM 1 Rejecting Calls (SIT Tone/Invalid) Across PRI
8:47AM 0 Newbie Here
7:51AM 1 outgoing MSN on zaphfc
7:33AM 3 bug or feature?
6:40AM 7 PostgreSQL
6:37AM 2 Anyone got latest SIP image for Cisco 7960?
6:26AM 0 Sound Distortion using IAX?
5:13AM 0 SV: dialing multiple extensions
2:27AM 0 act via Network or web?
1:22AM 5 cdr_odbc with mysql on a remote server
12:58AM 9 CTI (Computer-Telephony Integration) with Asterisk ?
12:49AM 0 NMI occures while loading the zaptel module
12:35AM 1 ISDN BRI Problem
12:28AM 4 Forwarding and record
Tuesday May 25 2004
9:42PM 4 Can I do this ...
8:36PM 1 CVS checkout problem
3:43PM 1 problem with vigor 2600v
3:39PM 0 No sound for MusicOnHold and SayDigits
3:37PM 1 No ringing on inbound DID calls
3:27PM 3 "Glare" condition - How well does asteriskhandle?
2:16PM 1 D-Channel on span 1 up/down + frame slips with zaptelBRI
2:07PM 0 MSN selection when dialout ISDN (ttyI* modem -interface, NOT CAPI)
1:01PM 1 SMS
12:52PM 6 Downgrading Asterisk
12:47PM 3 Voice Pulse
11:53AM 8 "Glare" condition - How well does asterisk handle?
11:46AM 1 FYI: Cisco firmware 7.1 released
11:35AM 4 79XX converting
11:25AM 0 Still Adtran and T100p
11:01AM 1 Call Admission Control
10:56AM 3 Telus: Overseas calling
10:48AM 0 asterisk and partner acs
10:40AM 1 Unable to create channel of type 'CAPI'
9:14AM 1 voicemail notify to external number
8:38AM 1 high pitched tone and message on answer
8:17AM 1 using asterisk with iptel addreses
7:51AM 1 (no subject)
7:23AM 1 dialout=fromvm
7:08AM 1 Zhone Zplex issues
6:48AM 0 Problems with IAX configuration
6:29AM 1 Problem - Adtran TSU 600, t100p
6:27AM 0 SMDi Interface?
6:13AM 4 fax/sandsp segfaulting asterisk
5:56AM 10 spandsp hylafax asterisk and confusion
4:51AM 1 SS7 links
4:38AM 0 Asterisk and Sipp
4:08AM 0 Compiling on OSX 10.3.3
3:44AM 1 Troubles with Kphone]
3:43AM 0 [Fwd: Answer App hanging in I4L]
3:30AM 1 Answer App hanging in I4L
3:15AM 0 Sound card problem
2:52AM 1 Troubles with Kphone
2:21AM 2 sip phone problem
2:07AM 4 Sip/IAX Clients for Linux
1:50AM 0 Question IAX and SIP bound to different IP's on the same * box
1:31AM 1 Speed Dials
1:21AM 1 SipTone II and Choppy/Stuttering Audio
12:20AM 1 Using Ser and Asterisk together
Monday May 24 2004
9:37PM 1 threewaycalling and #
7:43PM 1 no delivery from queue on IAX2 extension
7:18PM 6 11 instead of Star
7:13PM 2 Newbie extensions.conf I need to include [SMS] context.
5:20PM 3 100 analog phones?? HOWTO?
3:26PM 4 Grandstream message light button
3:11PM 5 mpg123
2:58PM 0 Asterisk connected to DataBase
2:55PM 1 Using Blacklist
2:25PM 2 SIP Authentication Problem
2:12PM 7 Sip Registration Problem
1:44PM 0 H.323, video and asterisk....
1:15PM 0 NCS support?
1:08PM 4 using the asterisk mailbox utility
1:00PM 2 testing asterisk on FXS lines
12:29PM 1 Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs
12:12PM 2 TerraCall Setting
12:00PM 1 CDR destination when user presses '#'
11:47AM 4 dialing multiple extensions
11:13AM 1 extensions/sip from database?
10:34AM 1 Channelized T1, SIP phones, HW Echo Canceller
10:19AM 2 IP local loop?
9:07AM 0 Help with IAX , voice Distortion or Breakage
8:41AM 1 Remote Disconnects
8:31AM 0 zapata ? question
8:21AM 3 Meetme Options (new one)
8:17AM 1 Fw: creating a single user voice mail box on asterisk?
8:17AM 1 Fw: setting the number of rings befor asterisk picks up?
7:35AM 0 ZapRas problems
7:17AM 1 Chan_capi 0.3.1 , Asterisk , 3 x C4 active ISDN card Segmentation fault
7:07AM 1 detecting pick up?
6:32AM 1 zaprtc-for-2.6
6:27AM 3 Help with IAX , voice Distortion or Breakage.
6:13AM 0 Variable AGI Parser
5:53AM 1 Cisco & Asterisk
4:58AM 1 SetVar - bellcode and cisco phone
4:00AM 0 IAX problems using CVS HEAD, but not CVS STABLE
2:03AM 0 Asterisk Audio Problem
12:54AM 0 updating the dial application
12:29AM 5 2 Sip phones behind un-natted Asterisk
12:19AM 1 Problem receiving a fax with RxFAX
Sunday May 23 2004
11:40PM 0 STREAM FILE question
10:46PM 2 Document
9:49PM 1 Aastra ADSI phone
7:52PM 5 PRI problem???
7:36PM 1 asterisk prompts?
6:19PM 0 setting the number of rings befor asterisk picks up?
6:05PM 0 Sipura SPA-3000 Beta
5:48PM 0 HELP!!! How do I move voicemail files to a new machine?
5:39PM 1 Serious NAT problems: can't call between lines on sipura
5:08PM 0 Fwd: regulating voip - aca
4:37PM 3 NetJet and RAS
4:10PM 4 Asterisk Prepaid
3:56PM 0 creating a single user voice mail box on asterisk?
12:45PM 0 ztdummy - how to test?
12:09PM 1 extension pattern matching
8:05AM 1 IAX2 NAT / Registration Issue
7:57AM 1 ZAPTEL not loading on FC2
5:22AM 1 *** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"
4:11AM 1 SIP with TerraCall Error
Saturday May 22 2004
11:21PM 1 Sip proxy registration help
7:45PM 0 CallerID and AON in Eastern Europe
7:40PM 1 Problems using Adtran 750 FXO and TE405P
6:24PM 1 Asterisk-oh323 0.6.1 Compiling problem
5:40PM 3
4:16PM 4 sip call using name in sip.conf
3:43PM 1 app_queue and app_groupcount
3:38PM 0 ast_rtp_read: Unknown RTP codec 72 received
3:24PM 14 Caller ID with BT CD50
3:11PM 5 Asterisk firewall config
2:50PM 0 HOW do I restore voicemail from backups?
1:44PM 1 Failure while compiling
12:32PM 1 Re: Sipura and STUN (was: rejected NOTIFY re quests)
11:56AM 1 Re: Sipura and STUN (was: rejected NOTIFY requests)
10:57AM 2 loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/ undefined symbol: ast_moh_stop
9:51AM 2 RxFAX generates no tiff file
9:49AM 3 fwd on busy when calling multiple extensions at once
9:34AM 0 asterisk cpu load
8:39AM 2 How to share Zap channels in 2 Asterisk servers
7:36AM 1 Dynamic SIP.CONF
7:18AM 2 rejected NOTIFY requests
5:36AM 1 call waiting indicator do not work for me.
3:24AM 2 Chan CAPI and Latest CVS wont compile
12:44AM 1 Is it Possible
Friday May 21 2004
7:29PM 4 G.729a beta codec on old Pentiums
6:39PM 6 VoicePulse SIP
5:21PM 1 Dumb TDM400P question
4:25PM 5 T100P HDLC configuration
3:25PM 4 dial application - continue in context
1:57PM 5 Making a SIP call
1:51PM 0 voicemail removal script
1:20PM 2 Asterisk upgrade on production box
11:07AM 0 Bridge calls
10:48AM 3 Asterisk and OH323
10:01AM 1 SIP Link on Web Pages
9:57AM 2 Failed to bind to Address already in use
6:20AM 0 Re: Your details
4:46AM 2 dial an IP address
1:40AM 4 Some problems with download Asterisk-addons
1:25AM 0 unable to use EXEC in AGI
Thursday May 20 2004
9:10PM 2 SwissVoice IP phones
8:21PM 3 UIP 200
5:19PM 0 Re: need phones with line and call indicators
4:47PM 4 Problem with SIP softphone
4:39PM 0 Variable value lost after call parking
3:46PM 8 I have put iLBC at the top
3:43PM 0 MSSQL2000 + cdr_odbc.c fix (WAS: problem with cdr_odbc)
2:42PM 0 Phone_read Resource temporarily unavailable
1:47PM 0 rtpmap issue w/Grandstreams
11:26AM 1 G729A problem
10:50AM 1 Avaya Partner Phones to SIP?
10:33AM 0 Re: My details
10:28AM 4 snom 200 and hold
10:28AM 3 Anonymous sip register
10:08AM 0 Error running festival command
10:04AM 2 asterisk-providers mailing list?
9:52AM 6 VoicePulse broken?
9:20AM 0 Time Limit Warning File
9:16AM 2 codec used on E1
8:58AM 6 G729 codec for asterisk
8:57AM 1 Premisys Slimline CB
8:40AM 4 Mystery SIP channels
8:30AM 0 budgetone problem on hangup
8:15AM 1 FC2 compile of zaptel
8:07AM 1 Mysql
7:54AM 1 Tellabs 2572 Configuration Advice?
7:44AM 4 x100p card + dailing out
6:39AM 1 Grandstream tftp cfg.txt format
6:18AM 1 Softphone Audio problem
5:47AM 1 voicemail customization
5:19AM 0 MultiTech MVP200 and Iconnect
4:59AM 1 Problems with Quadbri card
4:14AM 2 AGI/php script not working
2:57AM 0 DTMF problems to connect CME to Asterisk.
2:41AM 0 disa issue
2:12AM 2 Fedora Core 2 and Kernel 2.6
1:20AM 2 Softphone lag
Wednesday May 19 2004
10:19PM 1 Using stutter dialtone like the PSTN does
9:58PM 1 asterisk - cisco as5400 (h,323) - PSTN
7:42PM 2 MGCP error dialing
6:57PM 2 O'Reilly Open Source Convention in Portland
6:30PM 1 avoiding rtp triangle
4:48PM 2 CallCenter setup
1:52PM 1 voicemail notify problem on sip extension
1:41PM 0 problem with ignorepat
1:14PM 3 Remote Call Forwarding
12:58PM 1 I'm in ADSI hell...
12:46PM 2 how to pass call duration to an agi script
12:46PM 1 Swissvoice ip10: No 3-way-calling! (MGCP)
12:40PM 3 Call recording between SIP phones
12:19PM 0 H323 + asterisk + NETMEETING
11:46AM 1 One-way audio with H.323 --> SIP call
11:39AM 1 iconnect register problem
11:21AM 0 example of mulity company extension.conf needed.
10:51AM 4 TDM400P problems with 1 FXS, 1 FXO
9:52AM 1 xp100 not hanging up after call disconnect
9:44AM 1 Strange Sip (FWD, SipGate and such) problem
9:37AM 4 Asterisk External Ringing
7:49AM 1 Gotta love Ellen Muraskin (RE: OMG THE SKY IS FALLING!! NOT!!!)
6:55AM 0 Video support SIP and IAX2
5:58AM 2 persistant call variables
4:42AM 1 Old sound in new call.
3:41AM 1 verify Request URI
2:33AM 1 What has happened to my asterisk/PRI ?
2:12AM 1 FreeBSD + Zaptel + Asterisk
1:47AM 1 using iLBC
1:45AM 2 Problem compiling zaptel with BRIstuff 0.0.2
Tuesday May 18 2004
10:08PM 1 Linejack dialout
9:26PM 5 want to set a var in sip.conf
9:06PM 0 FW: * and Cisco routers
8:55PM 0 Asterisk not answering phone
8:23PM 0 Asterisk to IAXTel help
6:54PM 3 Free Softphone Recomendations
6:42PM 7 Asterisk on Compact PCI platform
6:19PM 3 call announce? in MeetMe?
5:50PM 2 My TDM-400P FXO experience
5:45PM 2 * and Cisco routers
5:05PM 1 Problem with QuadBRI
5:04PM 2 ADIT 600 Manual
4:48PM 0 tying a call to indications/call progress tones
3:00PM 0 using ast_request("zap", format, "pseudo")?
2:09PM 0 zaphfc Compile Error
1:31PM 1 how does a sip://user@dom.ain url come in
12:50PM 1 Asterisk on OS X
12:48PM 0 snom 200 phones.
12:35PM 0 MeetMe conference delay increasing
12:18PM 1 TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls
12:18PM 5 blocked caller id
11:28AM 5 AArgh, * and the 7960
10:38AM 0 PRI CID name
10:06AM 0 403 Forbidden since upgrading
9:01AM 1 How can I dial (0 + telephone number)
8:45AM 11 ATA devices
8:23AM 2 registering in sipphone
8:13AM 5 X100P answer in first Ring
7:43AM 1 G.729 on /dev/sda
7:24AM 2 problem with cdr_odbc
7:19AM 1 VoiceMailMain dumps user back into my incoming context after leaving a message
7:04AM 1 VoIP Termination w/ 402 or 712 area code?
6:41AM 2 asterisk voicemail retrieval using a cisco 7940
6:40AM 0 oh323.conf
6:27AM 0 problems with asterisk-oh323
4:46AM 1 R: Configure asterisk for outgoing.. need authuser parameter?
4:46AM 1 Dial and MeetMe on the same channel
4:19AM 0 No luck using asterisk as proxy...
4:03AM 1 Configure asterisk for outgoing.. need authuser parameter?
3:49AM 3 Q.931 clearing causes
2:50AM 1 DateTime bug?
2:04AM 0 Outbound call using Soft Phone
1:47AM 0 Number portability
12:35AM 0 Re: X100P Red Alarm Ireland
Monday May 17 2004
8:59PM 2 Problems w. chan_capi + ztdummy
8:27PM 0 failed compile
8:20PM 0 Corrupt Callerid Data
5:53PM 0 mgcp with busy tone
5:52PM 0 CAPI<->SIP broken incoming audio
5:20PM 0 *, Sipura, Call-Waiting, X100P, 2 ZAP Calls
5:13PM 0 Rate Engine Application
3:59PM 4 total newbie sanity check
3:07PM 1 problems compiling h323 support
2:33PM 0 Something weird
2:29PM 0 DTMF transmitted over IAX2 coming out as clicks at the other end
1:56PM 0 RE: question on domains requiring SRV lookups within asterisk
1:12PM 0 Astricon 2004 - the developer's meeting ** CALL FOR PAPERS
1:09PM 0 iax2 and ethereal
12:49PM 0 Zap callwaiting hookflash idiosyncracy/flaw?
11:51AM 4 *8 problem still there?
10:27AM 0 dial without answer ?
9:41AM 0 span_dsp & faxing: segmentation fault
9:35AM 4 Redhat 7.3 compiling problem
8:54AM 0 Snom200 Firmware: I only see 2.04g
8:41AM 1 Snom200 Firmware: I only see 2.04g
7:45AM 0 openbsd compilation fails for recent checkout of v1-0_stable
7:17AM 0 Cisco 7940/7960 users
6:10AM 0 Some thougts about implementing native 3-way calling and attended transfer
5:38AM 2 recommended hardware for quad E1 system
2:28AM 4 Asterisk Proxy Type
1:59AM 0 Tones...
12:52AM 2 Grandstream phone from speaker phone back to handset
Sunday May 16 2004
7:20PM 2 Re: say.c compilation error
7:08PM 0 Caller ID from Call Pickup
6:39PM 2 (no subject)
2:17PM 4 X100P Ireland Red Alarm (AR Tarzi)
1:06PM 0 CallerID information on H.323 channel
12:15PM 6 X100P problem with PSTN from BOLIVIA
9:51AM 1 Vertical applications?
9:03AM 0 Call forking/parallel call cdr.
7:21AM 2 cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes)
6:31AM 2 Snom200 ?
6:07AM 7 Grandstream v1.0.4.68 firmware
3:37AM 1 ** Asterisk Sunday Morning News: Contribute to the community
12:34AM 0 sip show channels
12:32AM 0 snoom200 and g729
Saturday May 15 2004
10:50PM 1 TDMoE hangs the machine
10:02PM 0 *8 (call pickup) using Manager or AGI interfaces?
7:43PM 1 110 extensions
7:11PM 1 Newbie question-no outgoing audio
4:20PM 4 Dial to Answer -- Can this be done?
3:21PM 0 Some doc
12:01PM 2 Subject: Re: X100P Ireland Red Alarm
11:43AM 1 Voicemail transfer
9:47AM 1 G729 Registration unsuccessful
9:13AM 1 Power alarm on module 1, resetting.
8:07AM 0 echocancelwhenbridged=no ?
5:15AM 1 asterisk with E1
1:17AM 1 X100P Ireland Red Alarm
Friday May 14 2004
11:26PM 4 IP-PSTN / PSTN-IP Gateway Service Providers
9:40PM 2 Data through T1, nethdlc
9:00PM 1 Dead FXO Module on TDM400P?
7:14PM 0 MGCP information
6:03PM 0 Bandwidth measurement tools (was: GSM v iLBC for low bandwidth connections)
5:04PM 4
3:46PM 3 SoftPhone to SoftPhone with No Voice
2:36PM 0 Interested on Sip to PSTN gateway
2:28PM 0 [Daniel Golding] Re: New VOIP Peering/Interconnection Mailing List Announcement
2:12PM 1 Loop length supported by FXS module on Wildcat TDM400 card
1:22PM 0 SuSE & Zaptel Compilation errors
12:32PM 3 X100P and TDM400P non-USA Caller ID
11:57AM 2 Fwd: [ISN] Voice Over IP Can Be Vulnerable To Hackers, Too
11:47AM 7 What's in ${EXTEN} ? Why does voicemail prompt for an extension?
11:47AM 4 How to Echo extension number to caller?
11:43AM 4 app_dbmysql and ODBC Voicemail
10:37AM 2 Scalable IVR
10:36AM 0 "skipping" / dropped audio, high-pitched squeal on X100P channels..
9:10AM 2 grandstream and timeservers
8:53AM 7 Business Looking Analog Phone
8:30AM 0 vovidia
8:28AM 4 snom 2.05b firmware
8:07AM 0 Overriding indications for IAX2 calls
7:44AM 1 Caller ID with NAME on PRI
6:50AM 0 TDM40P
4:31AM 2 GSM v iLBC for low bandwidth connections
4:25AM 0 Large Scale Deployments
3:51AM 0 Perhaps France isn't asleep also or a least part of it.
3:11AM 3 snom & gsm codec
2:53AM 2 Help needed with bri-stuff.0.02. slw91 k2.6.5
2:19AM 3 Psssst. The US is asleep - let's talk intern ationalization !!!
1:29AM 0 3 Q's about queues and agents
1:29AM 4 snom200 call wait indication
1:06AM 1 chan_capi broken incoming audio
12:26AM 1 Psssst. The US is asleep - let's talk internationalization !!!
12:10AM 4 sip authentication
Thursday May 13 2004
9:25PM 0 NI2 signaling with * for CID&NAME?
7:36PM 2 Asterisk, Configuration of SDP in SIP messages
5:16PM 0 Consult transfer on SNOM 105
4:38PM 1 pattern matching w/ Cisco dialplans
3:36PM 1 TDM11B - 2nd channel is not detected
3:09PM 0 (no subject)
2:57PM 1 MeetMe with AGI scripts
2:44PM 0 vocera
2:43PM 3 recommend a Linux based TFTP server
2:24PM 6 IAXy
1:40PM 4 IAX Freeworld
11:41AM 2 Can asterisk be programmed to make "alarm calls"?
11:34AM 0 ERROR[147466]: chan_capi.c:1914
11:25AM 0 ISDN & Voicemail: Strange Behaviour
10:44AM 1 poll vs select in channel.c
10:41AM 1 asterisk-doc Conference Call - phase 2 :)
8:28AM 0 IVR with chan_h323 module
8:22AM 0 T100P Hang up detection
8:15AM 2 Unable to play dialtone on channel xx (Zaptel TE405P)
7:57AM 4 BGM Music
7:48AM 5 Budgetone iLBC to IAX2 iLBC
4:39AM 0 Taiwan calling!!!!
3:07AM 1 DASS2 support
2:55AM 1 Where are the list archives??
2:20AM 0 Consultive Transfer, or faking it
1:44AM 0 extensions in mysql
1:41AM 0 zaptel does not compile on latest RHEL kernel
1:22AM 1 SNOM200 + 2.05a Firmware + G729b BUG
12:09AM 0 MGCP channel problem
Wednesday May 12 2004
9:47PM 2 Sonicwall with Firmware 6.6.02 - SIP?
9:07PM 4 Losing my PRI Interface every 20-30 minutes???
7:25PM 2 Newbie voiceplus + asterisk
7:23PM 2 iax behind a SonicWall
3:58PM 2 problems with analog interface to PBX
3:25PM 1 CISCO 30 VIP phone / 12 SP+ Connection does not free up
3:21PM 5 2.05a firmware
2:59PM 0 User Instructions for voicemail system
1:33PM 0 Simulating national-/internationalprefix ?
1:04PM 0 (no subject)
12:54PM 2 Simulating Dialtone ?
12:16PM 6 Dell server for asterisk question!
11:06AM 0 cisco 7960 line and extension apperance
11:05AM 1 Musical interruptions
10:18AM 1 G729 Segmentation fault
10:12AM 1 Asterisk Questions
9:54AM 2 847 IAX Provider??
9:46AM 0 Zaphfc ver 0.0.2
9:39AM 2 * and sip proxy auth
9:24AM 0 Chan_Capi & Modem/ttyI
8:53AM 0 quadBRI telco part hungs
8:42AM 3 Cisco 7960 SIP - DND soft key toggle?
8:03AM 1 Ring Tone - SIP / IAX2
7:26AM 1 Asterisk not loading data into table using cdr_odbc
7:13AM 2 cdr_mysql - would index slow down?
6:43AM 2 CDR-MySQL
6:37AM 0 Problems Retrieving Voicemail Remotely
5:08AM 3 Needed Open Ports
5:04AM 6 Where to get 48 volt Power Supplies for Cisco IP Phones
4:43AM 2 Calling CHRIS BARNET (PRI / E100P / ntl)
4:26AM 0 chan_CAPI uses Modem/ttyIx ?
3:42AM 0 [DTMF] Audio-Before-Answer issues
3:39AM 2 Re: [Asterisk-doc] Conference hosting request for asterisk-doc
3:19AM 0 fine mode receive fax problem
2:57AM 0 no sound for inbound calls
12:50AM 0 Asterisk Downunder (Australia & New Zealand)
12:41AM 0 SIP using h323 to gnugk
12:41AM 0 Voice Flow
12:18AM 1 Multiple ISDN controllers & Capi
12:04AM 2 Good source for Polycom IP Phones
Tuesday May 11 2004
11:49PM 1 Fax detection on IAX2 channel?
3:17PM 2 SDP messages relating to rtpmap Question
3:02PM 1 Caller-ID for alphanumeric SIP uris
1:21PM 0 midget packet received
1:07PM 0 Disabling agent call logging
12:27PM 1 Areski CDR graph incorrect
11:55AM 1 Need help: X100P connection/configuration in GERMANY
11:03AM 1 KIRK System 600 IP and Asterisk
9:55AM 1 Kernel Freezes with T100P
9:42AM 2 Asterisk + VoiceWorks
9:22AM 1 Sipura, Asterisk, *0, and Call Waiting
8:09AM 1 Long delays when talking betwwen SIP phones
7:58AM 0 New patch for Bug 1420
4:10AM 2 Sipgate to regular phones
3:15AM 3 Line appearances
3:07AM 1 Asterisk resource consumption..
2:52AM 1 Use buttons (other than #) after call is bridged?
Monday May 10 2004
10:52PM 1 wait_for_digit() not working for me!!!
9:26PM 2 Notice for Gentoo Users In Regard to mpg123
9:02PM 0 +5 seconds delay when Switching to macros
8:23PM 0 How do I catch someone pressing the * key?
7:43PM 1 Terrible TICKING sound
7:08PM 1 help-listening to my mailbox
6:37PM 1 ztcfg and Aastra 390 phones
6:06PM 0 CISCO 30 VIP and 12 SP+
4:58PM 1 DNS load-balancing & SRV records
4:47PM 0 polycom ip 500 registration problems
4:11PM 2 Asterisk on a dual processor machine
2:02PM 1 Callerid via PRI
1:16PM 6 SIP calls-per-second performance test tool
1:13PM 5 Swap partition/file and Asterisk.
12:33PM 0 Uniden UIP200 Review (Repost)
12:11PM 2 alternative FXO gateway to Mediatrix 1204?
11:22AM 3 Question about Asterisk and its use
10:53AM 6 Virbiage FT201 IAX Hard Phone
10:12AM 0 Re: bill
8:20AM 1 Dropped calles (with mp3)
8:10AM 1 News
5:58AM 1 mailbox numbers
5:53AM 2 Explain cidinternalcontexts?
5:47AM 1 Signalling C7 / SS7
3:44AM 1 app_sms - rocks!
3:27AM 3 Asterisk & Rhetorical Systems
3:27AM 2 problems compiling oh_323 and asterisk
2:39AM 0 SIP seeding
2:02AM 1 Problem with SMP?
1:53AM 1 Testing IP phone (g729, g711) with Windows Messenger (g723, g711)
1:33AM 0 SIP Error: Network Unreachable
Sunday May 9 2004
11:22PM 11 SIP in the UK
10:56PM 1 Example: calling card using extension logic ONLY!
8:54PM 1 Problems when upgraded
5:18PM 4 Asterisk webmin
3:51PM 2 Help with initial setup
2:28PM 3 German sound files available
2:09PM 3 AGI Assistance
12:50PM 0 Cannot Dial out with xp100
12:47PM 1 AGI Assitance
11:23AM 3 DTMF broken
9:47AM 1 No outbound calls at a PRI possible
9:39AM 1 Stripping numbers at the end of a dial pattern => extension
7:32AM 2 Sip to PSTN Gateway Configs
7:02AM 2 Help!! Music On Hold
6:20AM 2 ztdummy problem?!?
5:59AM 3 Where to start?
3:54AM 0 Telekom ISDN CFU is it possible ?
3:48AM 0 asterisk/can_capi took ISDN B Channels busy.
1:20AM 0 Re: Question
12:59AM 1 *** Asterisk sunday news: Read the sample configs, Luke!
Saturday May 8 2004
11:12PM 1 500ms usleep in rtp.c ?
3:03PM 1 Stripping numbers at the end of a dial pattern => extensions.conf
3:02PM 3 asterisk with german SIPGATE ?
2:10PM 2 x100p / Answer-> Flash -> Dial
1:51PM 0 H323 - Gatekeeper - asterisk - SIP config problems
1:10PM 5 1800 Provider
12:08PM 0 Failover Scenario - synchronizing voicemail & key files
11:09AM 3 Transfering with Grandstream Phones
9:46AM 2 List of online sip users
9:02AM 0 AVM B1 ISDN Call forwarding
7:41AM 0 need working loopstart config - t100p
3:20AM 0 authorise with h323 client at the * via gatekeeper
3:15AM 0 Indication Busy to a ZAP ISDN channel
Friday May 7 2004
7:45PM 0 Callwaiting callerid on 390s?
7:20PM 6 X100P keeping PSTN line Offhook
5:57PM 1 meetme conf-background.agi
3:57PM 3 MPG123 errors
3:38PM 0 RE: PRI, multi D channels and conventional PBXs (brian)
2:55PM 4 Concept for line appearances and bridging: anyone?
2:43PM 1 Voicemail: upgraded?
2:17PM 1 Uniden UIP200 Review
1:34PM 0 Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.64,1.65
1:13PM 5 729 licence on scsi
12:48PM 0 Manager, command action
12:03PM 2 Availability of T400P and E400P
11:53AM 1 cannot play sound files
11:52AM 0 - RE: Missing digits on TDM400P incomplete dial string - DTMF problem? - Email found in subject
11:50AM 0 Routing by called interface - Email found in subject
11:49AM 0 - Re: Routing by called interface - Email found in subject
10:56AM 0 CAPI & Gain
10:34AM 3 Routing by called interface
10:12AM 2 PRI, multi D channels and conventional PBXs
9:01AM 4 Cisco 7940 Phones as paging system?
8:08AM 0 DTMF Echo on H323 Channel
8:08AM 1 Trunk with CIRPAK
7:45AM 2 Newbie x100p install question
7:43AM 7 Asterisk and Cisco 7960 problems persist (for me, anyway)
7:27AM 0 caller id detection
7:25AM 1 Missing digits on TDM400P incomplete dial string - Email found in subject
6:14AM 4 SIP Wokflow diagram
5:59AM 7 WI FI IP phones??
5:48AM 2 zaptel.conf question
5:41AM 5 SIP: Trouble with "Moved temporarily" (302)
4:58AM 0 PSTN line tests
4:37AM 0 modem (56k) call to PSTN
4:01AM 1 Cisco 7940 microphone volume
3:52AM 2 Trouble compiling latest CVS
1:51AM 2 quadBRI & ISDN telephone
Thursday May 6 2004
10:33PM 1 sip + zap problem
7:09PM 0 AVM Fritz doesn't pick up call or make calls
4:52PM 3 mpg123 versions ?
4:03PM 0 Please help the new guy (the s extension)
2:09PM 1 polycom dialplan
10:27AM 0 Unable to find the source of the error: bad file descriptora
9:52AM 4 asterisk-oh323, new version 0.6.1
9:47AM 0 error on loading...
9:11AM 5 Fehler beim starten...
9:02AM 4 Playing GSM files in Windows
8:05AM 0 Syntax (2)
7:49AM 4 Cisco 7920 Image
6:31AM 2 Sending Tones after * connects
6:12AM 0 no incoming audio on outgoing sipcalls
4:56AM 3 Dial internal phones problem - zaphfc
4:46AM 1 Date time problems
2:46AM 7 sip traffic.
2:43AM 0 Problem in extensions.conf
Wednesday May 5 2004
8:27PM 0 CDR reporting GUI
8:27PM 3 Mediatrix 1204 (4x FXO)
3:46PM 0 (no subject)
3:03PM 3 sip.conf and SIP client host= not recognized in some cases
2:39PM 3 sip via tcp
1:52PM 8 No Audio from Hard Phone to SIP
1:44PM 0 Asunto: Re: Syntax
12:46PM 0 Lost my G.729 licenses
12:39PM 2 chan_sip and Digest realm
11:26AM 3 Problem with PRI and overlapped dialing
11:17AM 9 Correct format of sip registry
11:16AM 1 MySQL and VoiceMail again
10:37AM 0 determining pass-thru mode
10:04AM 1 Digits in a different language...
9:31AM 1 strange sip behavior (looping back to my own extension vm)
9:19AM 0 app_sms woes
8:06AM 1 Asterisk devel. - Mediatrix dtmf bug solved
7:43AM 1 PRI to PRI fax pass through
7:42AM 0 CAPI & Eicon Crash Asterisk
7:20AM 4 Cisco 7905 vs Cisco 7905-G
6:42AM 0 Missing: Red alarm event in Asterisk Manager Interface
6:20AM 2 connect a sub telefon system?
6:04AM 0 Asterisk Dialogic support
5:45AM 0 I can not register via sip to iptel or sipgate.
4:20AM 7 * & ISDN-BRI-PTP & DID & ISDN4Linux does not show incoming number
3:08AM 1 SIP Pick up groups
2:24AM 0 Problem with extension.conf
2:10AM 1 Early B3
2:10AM 1 Problem in Extension.conf
1:45AM 2 BUSY tone
1:11AM 2 183 Session in Progress
Tuesday May 4 2004
11:01PM 2 Adtran ta750 Configuration
9:09PM 0 Cli and the called number info
9:05PM 1 VoiceTronix V4PCI and Digium TDM40B together
8:24PM 5 ACD and/or CTI components for Asterisk
8:02PM 1 vonage sip url
4:11PM 4 mediatrix 1104
2:59PM 3 g.729 - licenses and opinions
1:24PM 7 stun server
11:45AM 1 0.7.2 debs
11:17AM 1 Error when loading wcfxo
11:13AM 2 Dial zap and music on hold
10:37AM 2 Can Asterisk support R2 signaling
10:30AM 1 multiplle isdn card
10:21AM 6 DSL vs X100P
9:42AM 1 Pots Extensions
9:32AM 3 Linux IAX client
8:30AM 0 Quality differences of codecs from PRI to SIP
8:21AM 0 Sip provider group
7:53AM 3 would it be possible to...
7:41AM 0 Help on legacy hardware.
6:51AM 1 Syntax
6:48AM 1 asterisk + NEC integration
6:41AM 3 Maximum retries exceeded problem...
5:25AM 2 Max TE410P card on an Asterisk
5:19AM 1 Asterisk and windows h.323 gatekeeper calling problems...
4:50AM 1 How does Norvergence do it ?
4:46AM 2 If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns
4:37AM 1 How to implement & configure agents
2:35AM 1 MGCP: Current CVS works for you?
2:10AM 0 Siemens cordless phone
1:06AM 0 Asterisk - no outband DTMF with Mediatrix
12:30AM 1 Probs with oh323 driver: audio only in 1 direction
12:15AM 0 Czech sound files
Monday May 3 2004
11:43PM 0 Asterisk E1 and Cisco as5300
9:35PM 3 RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?
8:59PM 0 A-B ok; B-C ok; A-C Crap
5:46PM 1 How do you close a VoicePulse "Connect!" account?
4:07PM 0 Pulse dial: outbound?
3:37PM 2 Number of Digium cards in one box...
1:42PM 0 Cisco 7905 and as5300 + Asterisk
1:05PM 1 SIP Call transfer with RTP transfer as well?
12:29PM 4 How does Novergence do it ?
10:41AM 2 ISDN WAN ISDN bridge possible?
10:26AM 1 quad fxo
10:08AM 1 Problem with new sipura firmware 1.0.35a
9:58AM 0 Open Source SCGP
8:58AM 0 txfax: "Trainability test failed"
8:28AM 1 Asterisk remains in the media path
8:04AM 2 zap x100p
7:45AM 1 Start recording during call by pressingbutton sequence
6:36AM 3 Error building asterisk-0.9.0
6:01AM 1 dialing a remote phone system and then entering an extension
4:28AM 2 Digital Line Distortion
2:59AM 1 Réf.: Re: Asterisk with UUI support ?
12:43AM 1 Asterisk & MGCP / NCS
Sunday May 2 2004
11:06PM 1 * Newbie installation advice
8:16PM 1 Re: Adit 600 FXO card (Jon Brandon)
7:17PM 1 Channel Bank - Vina T-1 Integrator
6:02PM 0 X101P problems
5:41PM 1 Adit 600 FXO card
3:42PM 1 Why don't I get a ringing sound?
3:25PM 1 module help?
2:53PM 1 Voicemail or voicemail2?
1:15PM 1 FXO line hum w/ Z-plex 10
1:00PM 2 Cisco 12SP+
12:54PM 0 chan_vpd patches
8:57AM 6 Simple SIP X-Lite Configuration Failing
7:37AM 1 no dial tone
5:24AM 2 Talking SIP to Vocal
1:39AM 4 iconnecthere behind NAT, strange deal
12:32AM 1 phonejack and linejack in the same system
Saturday May 1 2004
7:15PM 0 FW: clicks at beginning of call
5:50PM 1 Searching Archives (Basic SIP Configuration Problem)?
3:28PM 0 RE: [E164-discuss] RE: E164 updater Client
2:36PM 1 Grandstream Ringtones
2:08PM 4 New TDM04B 4-port FXO card problems
12:30PM 1 Grandspream & call parking
12:18PM 0 Reviewers Needed
11:04AM 1 Fax Detect problem (have consulted archives, wiki & irc)
7:10AM 0 h323.conf: multiple hosts per user?
5:59AM 1 dialing out to PSTN from SIP phones
1:58AM 1 Outbound Dialling on ISDN using CAPI - Individual Dial out Plans using msns
1:02AM 4 New ENUM service, what do you think?
12:25AM 0 Asterisk, festival, dropped calls